similar to: Incoming SIP calls as asterisk@...

Now this is really frustrating. Everything was working fine, and now it isn't ... I don't think I've changed anything that would affect this, but I guess you never can be too sure. My setup is as follows: SIP softphone (SJphone) connected to Asterisk running my Linux NAT firewall box. This is all on the internal network. Asterisk then dialing out through various means - SIP to

Hello everybody, I?ve been pulling my hair for a week now over a problem, and I really don?t know where to look anymore. Here?s my setup: There is an Innovaphone IP3000 VoIP gateway on the LAN (10.253.30.254). I can use it to send and receive calls from physical phones attached to it. I have setup Asterisk 1.0.3, with H323 and openH323, and on the same server I also setup GnuGK (10.253.30.1). I

Hi, I'm aware that incoming and outgoing calls are going fine when isdn channels are involved - caller id properly identifies calling party, so user can call back.... But how to properly handle this for iax, sip calls.... I have few questions : - BTW, what to type for instance in remote firefly to make standalone calls to Asterisk default context or particular extension ? - If I receive

Hello all, I'm having a problem with getting incoming callerid to a lan-connected phone. The Asterisk server is connected to the Internet, and a Grandstream BT101 phone on a lan interface: INTERNET ----(eth0) Asterisk (eth1) ---- Grandstream (192.168.1.51) The phone registers with the Asterisk server as ext 20. I can initiate and receive calls from the Grandstream phone fine. The

configuration ISDN BRI card : ISDN Olitec PCI 128 (hisax gazel) internet connection by ISDN 64kb/s dynamic IP nom de domaine registered to : dyndns.org avec ddclient to register IP par ddclient asterisk (on internet gateway) configuration pour ISDN BRI par virtual modems /dev/ttyI* (modem.conf) logical telephone SIP "SJPHONE" on 2 local stations "windows" (i don't succeed

All, I am connecting to a CS 1000 nortel PBX. I can call out, I have limited success with call in. I get debug traffic that a call is coming in but I get the message "Unable to create/find channel". I was expecting that incoming calls over the trunk would be handled from my sip definition and goto the nortel context. It is not. Below is the actual incoming call debug information. I am

Dear all, Mayybe this is a little off-topic but I don't know of any other place to ask for it... my apologies in advance! I'm looking for a softphone (SIP) with multiple profiles support. Right now I use SJPhone on SuSE 9.0 Pro, which allows to create several profiles but, AFAIK, it's not possible to use them all at the same time. I need this feature because I use different VoIP

I'm trying to figure out if this is a nat problem. I have a private network behind a freebsd nat box. The * server is on a static nat, with a private ip of 10.139.10.165. I'm connecting with sjphone as the client from 10.139.10.159. I am calling out using simpletelecom. When connecting directly to simpletelecom using sjphone everything works fine. When I go through * I get

Thanks a lot for all the suggestions! Unfortunately, it still gives problems. Most common error message is "ast_realaudio_callback Failed to write frame" after "paying the beep". Then it says "User disconnected". Also, it doesn't react to any extension entered and doesn't do any forwarding (as it should in "exten =>

Hello all, It's me again. I would like play with calls between a H.323 client and a SIP client through * inside my LAN. For that, on host 192.168.1.20, I use GnomeMeeting (GM20) and Asterisk; on host 192.168.1.25, I use SJphone (SJ25) as SIP client on Windows and I register into * with a username, no password. The 3 files oh323.conf, sip.conf, extensions.conf are in attachment. In the same

the caller is out his money anyway when you call any phone and voicemail kicks in, although i think on a payphone they give you a 2 or 3 second window to hang up. Suggest you implement i'm here / i'm away dialplan logic or set the do not disturb button that way when someone calls and the guy is away it hits voicemail right away and the caller can hear this and still have the 2 or 3

I was troubleshooting a problem with incoming calls to my VoicePulse Open Access (NOT Connect) numbers not coming in and I noticed the following in the SIP debug... >Found peer 'roamer1-vpoa' >Looking for s00****** in ivr-incoming Why are the calls getting sent to this weird "s00******" extension and not the usual "s" extension in context ivr-incoming as they

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",

Hi, I am using SJPhone here for testing ivr with Asterisk. I haven't seen any problem here yet. I have tried different things for that and finally got it working. I am not an expert to explain more about that, but here is the general section form my sip.conf. dont know whether it will help... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ;

I'm trying to get into the world of Asterisk in order to use the voicemail and autoattendat features (and more stuff later) with a Redcom switch. But, I've only started and haven't gotten to that yet. At this point my solitary goal is to talk to the autoattendant via an SIP phone (SJPhone). I've spent countless hours trying to find the documentation I need to accomplish my goals

Hi, You need to have completely replaced the Microsoft driver, because it doesn't support the headset or ctp Bluetooth profiles. This gave me fits! I followed the instructions at http://www.windowsdevcenter.com/pub/a/windows/2005/07/05/bluetooth.html and it works with both a Plantronics and a Motorola Headset, and I can answer calls with idefisk, eyebeam, x-lite, and kapanga. If you end

When incoming DATA call arrive on ISDN BRI, asterisk (zaphfc) recognise that this is DATA call, but answering anyway (playing IVR messages, etc...) How to stop that? I want that only VOICE calls are answered, and DATA/FAX to be ignored. (I'm using Asterisk 1.2.1 Brisftuffed 0.3.0-PRE-1f, ZapHFC) Log: -- Accepting data call from 'XXXXXXXX' to '3001' on channel 0/2, span 1

Hello, i have one question regarding incoming SIP INVITES. I have a testbed where i have 5 extnsions : 6001 - 6005 Domain : domainA.com Then i have configured a sip trunk, where my PBX registers to a foreign SIP Proxy. All is working fine, until following scenario: Incoming call from 6002 at foreignB.com (SRV exists,user also exists in pbx as extension, but different domain!) When i try this,

Hi, I have asterisk register two users (client-1, client-2) with a SIP proxy. I have the same two SIP client registered with asterisk. Now my dial plan setup is such that any call from client-1/client-2 is forwarded to the SIP proxy and the SIP proxy then takes the routing decision. Calls coming from SIP proxy will dial out the respective user. Asterisk is required to stay in the signaling as